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IEEE Speech coding Workshop

作者： Agarwal, T Kabal, P McGill Univ Dept Elect & Comp Engn Montreal PQ H3A 2A7 Canada

ISBN: (纸本)0780375491

Accurate linear Prediction Coefficient (LPC) estimation is one of the key requirements for low bit-rate voice coding. Under harsh acoustic conditions, LPC estimation can become unreliable. This results in poor quality of encoded speech and introduces annoying artifacts. This paper presents a two-branch speech enhancement preprocessing scheme for low bit-rate voice coders. This scheme consists of two parallel denoising blocks. One block will enhance the degraded speech for LPC estimation. Another block will increase the perceptual quality of the speech to be coded. The goal of this paper is to design the two-branch scheme. Test results show that the two-branch scheme can provide better perceptual quality compared to conventional one-branch speech enhancement techniques in noisy environments.

关键词： speech enhancement parallel two-branch speech enhancement linear prediction coefficient estimation voice coders noisy speech preprocessing signal denoising parameter estimation vocoders speech coding random noise linear predictive coding Vocoders Speech enhancement signal denoising speech coding random noise Parameter estimation Parallel Lines Speech

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Multi domain features for speaker independent word recognition

Multi domain features for speaker independent word recogniti...

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Asia-Pacific Conference on Circuits and Systems

作者： Dutono, T Watanabe, A ITS EEPIS Surabaya Indonesia

ISBN: (纸本)0780376900

Currently, speech recognition technology is in a mature condition, however, comparing with others applied technology, speech recognition technology still have limited daily live applications. One of the reasons is that the lack of our knowledge regarding the speech features. We performed word recognition experiments using features that is extracted by several different ways. We called them multi domain features. As shown in the results of recognition experiments, it is interesting that the multi domain parameters which represent phonemic features are very effective for the word recognition and operate to promote the goodness of the recognition in cooperation with the other conditions like a distance measure.

关键词： Cepstrum Dictionaries Euclidean distance Feature extraction Filter bank Jamming linear predictive coding Pattern recognition Speech recognition Vocabulary

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A packet loss recovery method using packet arrived behind the playout time for CELP decoding

A packet loss recovery method using packet arrived behind th...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Serizawa, M Ito, H NEC Corp Ltd Multimedia Res Labs Kawasaki Kanagawa 2168555 Japan

ISBN: (纸本)0780374029

This paper proposes a packet loss recovery method using a packet arrived behind the playout time for CELP decoding. The proposed method recovers synchronization of the filter states between encoding and decoding in the period following packet loss. The recovery is performed by replacing the degraded filter states with ones adaptively calculated from the late arrival packet in decoding. When the proposed method is applied to the AMR speech decoder, it improves the segmental SNR by 0.2 to 1.8 dB at packet loss rates of I to 10 116, where all the packet losses occur due to their late arrival. The subjective test results show that five-grade mean opinion scores are improved by 0.3 and 0.2 at a packet loss rate of 5% at speech coding bitrates of 7.95 and 12.2 kbps, respectively.

关键词： speech coding decoding linear predictive coding synchronisation packet loss recovery method CELP decoding synchronization recovery filter states late arrival packet AMR speech decoder segmental SNR subjective test results five-grade mean opinion scores speech coding code excited linear predictive coding playout time VoIP applications 7.95 kbit/s 12.2 kbit/s

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Improved spectral tracking using interpolated linear prediction parameters

Improved spectral tracking using interpolated linear predict...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Pereira, W Kabal, P McGill Univ Dept Elect & Comp Engn Montreal PQ H3A 2T5 Canada

ISBN: (纸本)0780374029

Conventional interpolation of linear Prediction (LP) parameters can provide a poor spectral match to the underlying speech signal for the intermediate subframes, In this paper, we present a method of modifying the interpolation endpoint LP parameters to improve the spectral tracking over all subframes. This 'warping' algorithm is based on minimizing the distortion between the interpolated LP parameters and those computed directly from the speech signal. The algorithm has been integrated into the Adaptive Multi Rate (AMR) speech codec. Our results show that this method enhances coder performance by smoothing out the LP parameter tracks and reducing coding distortion.

关键词： spectral analysis interpolation linear predictive coding speech coding tracking pattern matching speech codecs speech enhancement spectral tracking linear prediction parameters spectral match speech signal intermediate subframes interpolation endpoint parameters warping algorithm distortion minimization Adaptive Multi Rate speech codec AMR speech codec coder performance enhancement smoothing coding distortion reduction

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Research of G.723.1 Speech Coder

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微计算机信息 2002年第8期 72-73+75页

作者： College of Computer & Comniu. Engi. Southwest Jiaotong Univ., Sichuan ?Chengdu 610031 Zhang Shaohua and Li Chengzhong College of Computer & Commu. Engi. Southwest Jiaotong Univ. Sichuan. Chengdu610031

This paper introduces concept of G.723.1 speech coder and analyses its technology and features. We advise to optimize its running time of G.723.1 speech coder. We put forward to improve some modules with large computa... 详细信息

关键词： speech coder linear predictive coding optimization

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iLBC - a linear predictive coder with robustness to packet losses

iLBC - a linear predictive coder with robustness to packet l...

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IEEE Workshop on Speech coding Proceedings

作者： S.V. Andersen W.B. Kleijn R. Hagen J. Linden M.N. Murthi J. Skoglund Department of Communication Technology Adborg University Denmark Global IP Sound San Francisco USA

In this paper, we discuss the internet low bit rate codec (iLBC) with an emphasis on the frame-independent long-term prediction. The frame-independent long-term prediction is a method to exploit pitch-lag correlations in the encoding of speech without suffering multiple-frame speech degradation in connection with transmission loss. We present mean opinion scores for the iLBC codec and show by means of signal examples how the nature of degradation in a predictive codec based on frame-independent long-term prediction differs from that of traditional CELP codecs.

关键词： Robustness Codecs Bit rate linear predictive coding predictive coding Decoding Internet Speech coding Acoustic noise Degradation

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Phoneme classification using naive Bayes classifier in reconstructed phase space

Phoneme classification using naive Bayes classifier in recon...

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IEEE Digital Signal Processing Workshop

作者： Jinjin Ye R.J. Povinelli M.T. Johnson Department of Electrical and Computer Engineering Marquette University Milwaukee WI USA Department of Electrical and Computer Engineering Marquette University Milwaukee WI

ISBN: (纸本)0780381165

A novel method for classifying speech phonemes is presented. Unlike traditional cepstral based methods, this approach uses histograms of reconstructed phase spaces. A naive Bayes classifier uses the probability mass estimates for classification. The approach is verified using isolated fricative, vowel, and nasal phonemes from the TIMIT corpus. The results show that a reconstructed phase space approach is a viable method for classification of phonemes, with the potential for use in a continuous speech recognition system.

关键词： Speech recognition Hidden Markov models Cepstral analysis Heart Delay effects Histograms Speech analysis linear predictive coding Neural networks Spectral analysis

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Implementation of speech recognition algorithm for a 32-bit CPU-based portable device

Implementation of speech recognition algorithm for a 32-bit ...

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IEEE International Conference on Consumer Electronics (ICCE)

作者： Suhong Ryu Younim Lee Wonyong Sung School of Electrical Engineering Seoul National University Seoul South Korea

ISBN: (纸本)0780373006

We have implemented a connected word speech recognition algorithm for a handheld educational device, SpeakingPartner. This system is based on a low-cost 32-bit CPU, and equips a graphic LCD for animation, 8 Mbyte of DRAM and 32 MB of NAND flash memory. Several previously known software optimization algorithms, such as fixed-point conversion, loop fusion, loop unrolling, and circular addressing, as well as efficient log-likelihood computation and partial backtracking methods are applied. The implementation results show that the peak computation requirement for a real-time implementation is reduced to about 8 MIPS, which translates that a 60 MHz 32-bit CPU based system can perform the speech recognition while playing animation in a multi-tasking mode.

关键词： Speech recognition Signal processing algorithms Animation Cepstrum Delay Floating-point arithmetic Random access memory Real time systems linear predictive coding Feature extraction

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Design and development of speech-control robotic manipulator arm

Design and development of speech-control robotic manipulator...

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International Conference on Control, Automation, Robotics and Vision (ICARCV)

作者： S.-H. Salleh Hong Kai Sze Tan Tian Swee Digital Signal Processing Group Faculty of Electrical Engineering Universiti Teknologi Malaysia Malaysia

This paper presents a speech-control robotic manipulator arm via personal computer and self-built robotic arm. It describes the software we developed to collect, process and train the speech database. First, the specifications of the robotic arm will be explained. The algorithms involved will also be briefly discussed. Finally, the result will show the ability of this system to control robotic manipulator arm with human's voice.

关键词： Manipulators Speech recognition Robot control Service robots linear predictive coding Microcomputers Speech processing Humans Industrial control Virtual manufacturing

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A new developed speech recognition education software in teaching LPC and LPC-derived cepstrum

A new developed speech recognition education software in tea...

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International Conference on Signal Processing Proceedings (ICSP)

作者： Hong Kai Sze S.-H. Salleh A.Z. Sha'ameri Digital Signal Processing Group Faculty of Electrical Engineering Universiti Teknologi Malaysia Malaysia

Speech processing systems are highly complex and teaching students in this subject matter with the underlying technologies can be a challenging task. Thus education software is needed to provide the students with more convenient way to study, perform experiments and develop his/her first speech recognition application without an instructor. The Speech Recognition Virtual Classroom (SRVC) was developed as education software in teaching speech recognition. There are various modules, like signal analysis, dynamic time warping, vector quantization and hidden Markov models (HMM). These modules are integrated into a single environment that allows easy learning through experiments with real speech data.

关键词： Speech recognition Education linear predictive coding Cepstrum Hidden Markov models Speech processing Software performance Application software Signal analysis Vector quantization

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